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  Unnatural reverberationUSPTO Application #: 20060086237Title: Unnatural reverberation Abstract: An electronic reverberation system employs a processor to produce a plurality of delays samples that are added to a direct signal to produce reverberant sound. The disclosed system generates or employs a list of gain value pairs that are produced based on control settings or are provided as fixed coefficients. The processor generates reverberation samples by applying these coefficients to delay samples and summing their amplitudes to produce reverberation waveform samples. The reverberation waveform samples are added to the direct signal. (end of abstract)
Agent: Weingarten, Schurgin, Gagnebin & Lebovici LLP - Boston, MA, US Inventor: Richard S. Burwen USPTO Applicaton #: 20060086237 - Class: 084630000 (USPTO) Related Patent Categories: Music, Instruments, Electrical Musical Tone Generation, Data Storage, Digital Memory Circuit (e.g., Ram, Rom, Etc.), Expression Or Special Effects (e.g., Force Or Velocity Responsive, Etc.), Reverberation The Patent Description & Claims data below is from USPTO Patent Application 20060086237. Brief Patent Description - Full Patent Description - Patent Application Claims
CROSS REFERENCE TO RELATED APPLICATIONS [0001] This application claims the benefit of Provisional Application No. 60/622,294 filed Oct. 26, 2004 and titled UNNATURAL REVERBERATION. STATEMENT REGARDING FEDERALLY SPONSORED RESEARCH OR DEVELOPMENT [0002] Not Applicable BACKGROUND OF THE INVENTION [0003] The present invention relates to audio systems and more specifically to an improved method and apparatus for providing reverberation. [0004] A listener in a room hears a combination of direct sound emanating from the sound source and a series of reflections from the room surfaces, which occur at different times. The frequency response at the listener location contains many peaks and valleys due to comb filtering, as all of the reflections and direct sound add together vectorially. Early attempts at electronic reverberation used a loudspeaker and microphone in a non-absorbent room. Later, space was saved by replacing the room with a metal plate or springs. When electronic analog delay became available, a decaying train of pulses could be produced by recirculating the output back to the input at slightly reduced gain. The development of computation and analog-to-digital and digital-to-analog converters allowed the same decaying train of analog pulses to be produced in the digital domain. [0005] Reverberation can be characterized by its impulse response. Mathematically convolving a music signal with this impulse response produces the reverberant signal. Therefore, development in reverberation has focused on obtaining a desirable impulse response. The latest method of producing electronic reverberation, now becoming popular, is to use sampling. Recording the impulse response of a concert hall and feeding it into a convolver makes a non-reverberant music source sound somewhat as though it was produced in that concert hall. [0006] Because of the large dimensions of concert halls, sound absorption by the audience and surfaces, and the speed of sound at approximately 1090 feet per second, listeners in even the best concert halls hear the direct sound at least 15 milliseconds before the first significant surface reflections arrive. The extreme high frequency content of the reflected sound is greatly attenuated relative to the direct sound. At low frequencies, depending upon the seat location, the reverberant sound usually exceeds the loudness of the direct sound. Some people like singing in a ceramic tiled shower stall where the reflected sound arrives much sooner and has more high frequency content. [0007] Electronic reverberation systems used in modern recordings have similar characteristics and provide more than 15 milliseconds of initial delay and attenuated high frequencies. The delay and lack of high frequency content in either the acoustic or artificial reverberation allows any noises or imperfections in the direct sound picked up by the microphones to be clearly heard. [0008] Most people do not realize they are listening to a beat frequency that occurs among multiple instruments or voices sounding the same note. Depending upon frequency, phase, and harmonic differences, a listener may hear a shimmering effect or high frequency noise. In addition, bowed instruments produce mechanical noises, and wind instruments produce wind noises and occasionally annoying high harmonics. Percussion instruments have rattles, and voices can be raspy on certain notes. Close microphone techniques often exaggerate these imperfections. [0009] Recording, transmission, and reproduction equipment may contribute their own imperfections or exaggerate those already present. For example, some recording engineers dislike the normal pulse code modulation (PCM) recording process due to irritating high frequency components they believe are not present in the live microphone signal. Lossy bit compression systems like MPEG-3 are also believed by some recording engineers to distort sound quality. Processes generally accepted by these same engineers are old-fashioned analog tape recording and new Direct Stream Digital (DSD) recording used in making Super Audio Compact Discs (SACD). Instead of 16-bit PCM at 44.1 kHz used for compact discs, DSD is 1 bit PCM at 2.7 MHz. [0010] Regardless of the sources of high frequency imperfections, the cumulative result is that nearly all existing recordings contain moments when the high frequencies are irritating enough to cause the listener to turn the volume down below the point of maximum enjoyment for the rest of the program--and sometimes to just turn it off. High frequency irritants can be reduced by attenuating the high frequencies using an equalizer. However, attenuating the high frequencies enough causes unsatisfactory loss of high frequency detail. [0011] It would therefore be desirable to have a system and method for reducing the imperfections, distortions and/or irritating effects exhibited by recorded material. BRIEF SUMMARY OF THE INVENTION [0012] In accordance with the present invention, a method and apparatus are disclosed for reducing imperfections in recorded material by means of improved artificial reverberation. The presently disclosed system produces smooth, non-irritating high frequency sound without sacrificing high frequency detail or creating a hollow sound. [0013] More specifically, the disclosed system, receives a series of digitized input waveform samples (known as the dry or direct signal) and temporarily stores each input waveform sample in a circular delay line having a predetermined number of delay line positions. The delay line is conceptually a First In First Out (FIFO) buffer. The delay line may be implemented as a circular delay line in a computer memory or a FIFO if implemented in hardware. A computational component utilizes a list of gain value pairs to create a reverberation signal including a series of reverberation waveform samples, each sample having an associated amplitude. Each gain value pair includes a first value that identifies a position in the delay line relative to the current sample position and a second value that specifies a gain coefficient. [0014] Each reverberation sample is calculated in real time by the computational component. To calculate a current reverberation sample, the computational component accesses each gain value pair in the gain value pair list. For each gain value pair, the computational component computes an intermediate value by accessing a prior input sample amplitude from a relative delay line position specified by the first value in the respective gain value pair and by multiplying that amplitude by the second value, or gain coefficient, in the respective gain value pair. The computational component calculates an intermediate value by performing this multiplication for each delay line position specified in the list of gain value pairs and adds all of the intermediate values to produce the current reverberation waveform sample. The reverberation signal is a series of reverberation waveform samples (known as the wet signal). [0015] A composite digital audio signal consisting of a series of composite waveform samples having respective sample amplitudes is generated by attenuating each current reverberation waveform sample and adding the attenuated reverberation waveform sample to the current input waveform sample. [0016] The lists of gain value pairs may be generated in several ways. In one embodiment, an operator sets a number of controls that establish certain parameters used to generate the list of gain value pairs. The computational component accesses the parameters and calculates the gain value pairs based upon the control settings established by the user. If the control settings are changed, the computational component generates a new list of gain value pairs based upon the new control settings. Since the adjustment of the control settings results in a modification of the list of gain value pairs used to generate the reverberation signal, the operator can adjust the characteristics of the reverberation signal via adjustment of the controls. [0017] In another embodiment, the reverberation component generates the reverberation signal using a pre-generated list of gain value pairs. One or more pre-generated lists of gain value pairs that produce varying reverberation signal characteristics can be provided. In the circumstance where plural lists of pre-generated gain value pairs are available, the operator is provided the ability through an interface to select which of the plural lists of gain value pairs is to be used to generate the reverberation signal. [0018] The first and second values in the list of gain value pairs describe an attenuation curve that includes a leading edge portion, a flat portion and a decay portion, the first values defining an X axis value and the second value defining a Y axis value. Parameters associated with these portions of the attenuation curve may be adjusted via the operator controls when such controls are employed. [0019] Unlike conventional reverberation systems, the list of gain value pairs in certain embodiments includes an initial gain value pair having a first value that specifies a delay line position that is delayed from the current time by a period of less than 15 milliseconds. The first values of additional gain value pairs in the gain value pair list may also identify delay line positions having delays from the current time of less than or equal to 15 milliseconds. [0020] In many useful waveforms, the reverberant energy is less than that of the direct sound at low and middle frequencies and gradually increases to where it exceeds the direct sound at very high frequencies. The reverberant energy does not necessarily increase per se at high frequencies. It can exceed the direct sound if the direct sound is attenuated as frequency increases. Continue reading... Full patent description for Unnatural reverberation Brief Patent Description - Full Patent Description - Patent Application Claims Click on the above for other options relating to this Unnatural reverberation patent application. ###  How KEYWORD MONITOR works... a FREE service from FreshPatents1. Sign up (takes 30 seconds). 2. Fill in the keywords to be monitored. 3. Each week you receive an email with patent applications related to your keywords. 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